Quote:Which is why digital audio is reframed every frame to the internal oscillator. Again, the part I'm unclear on is how the jitter is supposed to let the stream run long or short with regards to time. If you could shed some light on that - the actual practicality of what occurs to disrupt the waveform - do you suggest that some bits pass other bits on the wire (or in the fibreoptic bundle) to arrive at the receiver before others.Bren R.

I had to do a little more research to get a better explanation.

Here is a good link which describes the problem and presents the solutions. Professional equipment uses a double phase lock solution, which requires a seperate clock sync line.

As it turns out clock information is in fact transported with the data along an spdif connection. The problem is that the way it was designed, the clock sync information is flawed. What happens is the data is assembled and passed to the spdif transmitter which then grabs the clock info and inserts it into the data stream. He says that due to a structural problem with spdif (hence the stigma of spdif being associated with jitter), the data and the clock sync get messed up.

The first attempts at solving this problem, but also moving into very expensive equipment, the DAC needs to provide it's own internal sync which it then syncs up with the clock from the data (inserted directly into the stream rather than pulled directly by the transmitter). Alternatively, some designs allow for a seperate clock line. This of course means you need a transport that supports clock out and a DAC that supports the input.

In short, spdif by itself is not providing a proper clock. If you have an expensive DAC, it can resync the data with it's own clock. He also claims that cheap timing crystals have been proven to not be accurate enough for high end audio.

Noit was an ensoniq vxd-sd ? workstation.I bought it in the summer of 1990.first real purchase I made after getting a good job.

I can play and read a little, but my timing is poor. I guess you could say my 'jitter' is noticable. I sold it a long time ago at the pawnshop. Long sad story I will spare us all the details. Now I have a cheap usb powered M-audio unit. I really havent played with it much with the move and everything else this year.

If I was to buy a 'real' music keyboard, I would get a roland now. I think they have the best sound. although I really dont keep up with what is happening.

I didnt take many pictures back then, but I am glad I have a picture of that rack. I spent hours tweaking and reconfiguring that stuff. Lots of good times, I was the soundman for a band.

I'd be running sound and an attractive chick would come up. I'd think , finally, I was about to be recoginized me for my studleeness. And then she would say, " Is the drummer married"

Quote: He says that due to a structural problem with spdif (hence the stigma of spdif being associated with jitter), the data and the clock sync get messed up.

He elaborates more about this on the second page I linked. Basically the way the data is encoded it relys on certain voltages to represent those states. What happens is there are some instances where if too many 0's follow each other there is a delay before a 1 can be represented and this messes up the clock information. It's quite a bit more complicated than that and my backround is in CS not electrical engineering.

In any case after reading through his pages, there are lots of places where jitter gets introduced. The bottom line is, it's impossible to get 100% time synced data (since no timing crystal exists with that kind of accuracy). From the original recording, all the way through to playback you will be picking up jitter.

So back to the all important question. Does this really make a difference? Here is what he has to say about that.

Quote: In my personal experience, and I would dare say in common understanding, there is a huge difference between the sound of low and high jitter systems. When the jitter amount is very high, as in very low cost CD players (2ns), the result is somewhat similar to wow and flutter, the well known problem that affected typically compact cassettes (and in a far less evident way turntables) and was caused by the non perfectly constant speed of the tape: the effect is similar, but here the variations have a far higher frequency and for this reasons are less easy to perceive but equally annoying. Very often in these cases the rhythmic message, the pace of the most complicated musical plots is partially or completely lost, music is dull, scarcely involving and apparently meaningless, it does not make any sense. Apart for harshness, the typical "digital" sound, in a word.

In lower amounts, the effect above is difficult to perceive, but jitter is still able to cause problems: reduction of the soundstage width and/or depth, lack of focus, sometimes a veil on the music. These effects are however far more difficult to trace back to jitter, as can be caused by many other factors.

I respectfully disagree with you about the Rolands, at least if you're talking piano sound. I find the Yamahas, Kawais (just a bit of bias on this one. ), and Viscounts to be far superior. I don't know about effects sounds, though.

thanks for the reply, I dont feel slighted.what prompted me to say that is 17 yrs ago i got that ensoniq. it has a built in sequencer and aftertouch and all kinds of goodies i have long forgotten, but it didnt sound lush. Another guy got a roland, and another got a yamaha , and both sounded much better. I'm sure I'm outa touch

Well, a few issues immediately spring to life with this. First is it comes from TNT-Audio, a "tweakers" site, while not as huile de la serpente as some (ooh, cable risers!), the guy's drunk the Kool-Aid and tied on his sneakers.

By the third diagram, he's already off the mark - AES/EBU does not rely on a separate connection for clock, the "pros" don't send the clock on a separate wire - just like SPDIF, it's embedded in the single data stream. I'm sure someone probably has a proprietary way of sending the clock discretely, but I've never seen it done.

If we jump past his half dozen ways to clean his clock (pun intended) he finally gives hard information about how awful this poor mistimed signal has become... even at the worse he claims (30ns - 15 times more than the worst DIYAudio measures) that equals a mistiming of 0.00000003 seconds... a single sample of CD program audio only lasts 0.000023 seconds, so my hand-creation of jitter was actually 7.5 times worse than a theoretical "worst case" scenario, and was still undetectable.

Again, correct me if I'm wrong, I sometimes misunderstand the intent of some of these articles, but the complaint is that the signal can get as sloppy as 3.0 x 10^(-8) seconds one way or the other? I sincerely hope I've misread that... if I haven't, perhaps someone could suggest to him that a tenth of a point of relative humidity or degree of ambient temperature in his room would stiffen or loosen the grease on the CD/DVD drive spindle creating a much larger problem than that.

Quote:In my personal experience, and I would dare say in common understanding, there is a huge difference between the sound of low and high jitter systems. When the jitter amount is very high, as in very low cost CD players (2ns), the result is somewhat similar to wow and flutter, the well known problem that affected typically compact cassettes (and in a far less evident way turntables) and was caused by the non perfectly constant speed of the tape: the effect is similar, but here the variations have a far higher frequency and for this reasons are less easy to perceive but equally annoying.

Wow and flutter on cassette tapes I will give him. He says turntables are far less susceptable to W&F than cassettes (and usually 0.15% is good for a turntable, belt or direct drive) meanwhile my CBC Radio Reference Disc suggests that a "pass" for a CD player is less than 0.002%... doing that math, a "passable" CD player exhibits 75 times less W&F than an excellent turntable. This is equally annoying? To listen to, or when you have to admit that your audio system isn't nanosecond perfect?!?

Quote:Very often in these cases the rhythmic message, the pace of the most complicated musical plots is partially or completely lost, music is dull, scarcely involving and apparently meaningless, it does not make any sense. Apart for harshness, the typical "digital" sound, in a word.

Oh, god... melodramatic much? The music is dull and lifeless because of a 0.00000003 second "swing" in sample timing? Gracious, this guy must be a hit at the symphony... "that bassoon player ruined it for me, I could tell he drank mineral and not bottled water before the concert, his reed was vibrating 2 nanoseconds behind the alto sax!"

I was more interested in finding out what was supposed to be wrong with spdif. If we start with the assumption that a proper clock sync is needed and you don't get it all the time (supposedly it's data dependent, therefore sometimes it's correct and sometimes it's not), then there has to be jitter.

I didn't get the impression he was trying to sell me anything. The lessloss people were, but I'm not spending $2000 to reduce jitter, nor would I suggest anyone else do that.

What I understood the 2nd guy as saying, was that if you go out and buy a $20 CD player it will be using a cheap and unreliable crystal. Avoiding the bottom end players doesn't sound like bad advice to me. As far as DACs go, there are some relatively inexpensive ones (I don't have models or pricing) that do their own syncing.

In any case we went down this road because I contend that making random data changes to a wav file is not the same as jitter. I will agree that if you made changes to almost every sample (not just say 10 changes) and no one heard a difference, then they probably wouldn't notice jitter either. Still, these are 2 different things being tested.

I don't know if anyone can really notice or hear jitter. This can all be factual, yet at the same time it could also be meaningless if even with crappy crystals, it's undetectable by the human ear.

The proper way to test this would be to set up a DAC that allows you set parameters for it's timing. This way you could force it to deviate by some fixed amount from the clock. Once you have this, you would run ABX tests and see how much deviation from the clock is needed for differences to be heard. If it turns out to be a value that is much greater than even the cheapest equipment puts out, then we know this whole issue is BS.

Quote:I was more interested in finding out what was supposed to be wrong with spdif. If we start with the assumption that a proper clock sync is needed and you don't get it all the time (supposedly it's data dependent, therefore sometimes it's correct and sometimes it's not), then there has to be jitter.

I always assumed the jitter complaint was that a SPDIF frame was actually bolloxed by jitter, his article shows the complaint is actually much smaller than that - the information is right, it's just "marred" by timing issues. I'm not sure that's an issue we'll ever get away from. As long as there are moving parts in a player's transport (CD/DVD drive spindles, turntable platter motors, hard drive platter motors) there will be some wow and flutter. As long as data is synced to an oscillator, there will be timing errors. I think, like with THD levels in new solid-state amplifier circuits, we're at a subsonic level with the current technology... the weak link's proving to be our own hearing.

Quote:What I understood the 2nd guy as saying, was that if you go out and buy a $20 CD player it will be using a cheap and unreliable crystal. Avoiding the bottom end players doesn't sound like bad advice to me.

Another frame of reference for CD audio playback requirements is that a $30 computer CD-ROM can pass data reliably at up to 48x what is required for CD audio. For data. Which has to be bit-perfect. If you want to skip to a certain comfort level on the price tree, that's understandable, but you can stop at any level that has the features you want.

Quote:In any case we went down this road because I contend that making random data changes to a wav file is not the same as jitter. I will agree that if you made changes to almost every sample (not just say 10 changes) and no one heard a difference, then they probably wouldn't notice jitter either. Still, these are 2 different things being tested.

At this point, knowing the complaint about it, I wouldn't have to damage any data. I was always under the assumption that there was an expected loss/destruction of a single 44.1/16/Stereo frame of audio. They don't even contend that much, just that the timing isn't perfect to the nanosecond... that they measured timing differences of 2 billionths of a second one way or the other during playback. I'd point to my own testing here that if no one could hear noise inserted lasting roughly 2 one-hundred thousandths of a second, that this is probably inaudible.

For more perspective - it takes an electrical (audio) signal about 6.6 nanoseconds to travel through a 6 ft cable or a flash to travel through a TOSlink cable the same length.

Quote:The proper way to test this would be to set up a DAC that allows you set parameters for it's timing. This way you could force it to deviate by some fixed amount from the clock. Once you have this, you would run ABX tests and see how much deviation from the clock is needed for differences to be heard. If it turns out to be a value that is much greater than even the cheapest equipment puts out, then we know this whole issue is BS.